Speech coding

About: Speech coding is a research topic. Over the lifetime, 14245 publications have been published within this topic receiving 271964 citations.

System and method for overlapping audio elements in a customized personal radio broadcast

Abstract: A method for overlapping stored audio elements in a system for providing a customized radio broadcast. The method includes the steps of dividing a first audio element into a plurality of audio element components; selecting one of said audio element components; decompressing the selected audio element component; selecting a second audio element; decompressing the second audio element; mixing the decompressed audio element component with the decompressed second audio element to form a mixed audio element component; and compressing the mixed audio element component to form a compressed overlapping audio element component. The compressed overlapping audio element component may replace the selected audio component. The first audio element may be a song, while the second audio element may be a DJ introduction. Accordingly, the compressed overlapping audio element may be broadcast followed by the remaining components of the song audio element.

Conference bridge processing of speech in a packet network environment

Abstract: There is provided a conference bridge or transcoder configured to intelligently handle multiple speech channels in the contest of a packet network, wherein various speech channels may adhere to variety of speech encoding standards. For example, the conference bridge establishes framing and alignment of multiple incoming speech channels associated with multiple participants, extracts parameters from the speech samples, mixes the parameters, and re-encodes the resulting speech samples for transmission to the participants. In one aspect, a speech processing method comprises decoding a first bitstream according to a first coding scheme to generate first speech samples and a first side information; generating second speech samples and a second side information using the first speech samples and the first side information, for use according to a second coding scheme; and creating a second bitstream, encoded based on the second coding scheme, using the second speech samples and the second side information.

System for adaptively reducing noise in speech signals

Abstract: A method and system are provided for adaptively reducing noise in frames of digitized audio signals that may include both speech and background noise. Frames of digitized audio signals are processed to determine what attenuation (if any) should be applied to the current frame of digitized audio signals. Initially it is determined whether the current frame of digitized audio signals includes speech information, this determination being based upon an estimate of noise and on a speech threshold value. An attenuation value determined for the previous audio frame is modified based on this determination and applied to the current frame in order to minimize the background noise which thereby improves the quality of received speech. The attenuation applied to the audio frames is modified gradually on a frame-by-frame basis, each sample in a specific frame is attenuated using the value calculated for that frame. The adaptive noise reduction system may be advantageously applied to telecommunication systems in which portable radio transceivers communicate over RF channels because the adaptive noise reduction technique does not significantly increase data processing overhead.

Missing packet recovery techniques for low-bit-rate coded speech

Abstract: Since missing-packet recovery techniques for conventional PCM speech are not applicable to packetized speech communication systems with low-bit-rate coding schemes, quality degradation mechanisms are presented for missing-packet recovery techniques. These mechanisms are least significant bit (LSB) dropping, waveform substitution, and odd-even sample-interpolation schemes. A comparison of these techniques in terms of signal-to-noise ratio and perceptual distortion under packet loss conditions shows that the LSB-dropping scheme with embedded coding is the most promising technique for recovering missing packets. >

Codebook tables for encoding and decoding

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.